Linux-audio-dev June 2013

linux-audio-dev@lists.linuxaudio.org

33 participants
23 discussions

by Frank Kober

Hi, QMidiArp 0.5.2 has just seen the light of the day. It brings mainly two improvements. One is a comeback, that of tempo changes on the fly, and that now includes also tempo changes of a potential Jack Transport master. Also the Jack Transport starting position is finally taken into account, so that QMidiArp should be in sync also when starting the transport master not at zero. The second one is Non Session Manager support, mainly thanks to the work done by Roy Vegard Ovesen! Note that for compiling in NSM support you will now need liblo as dependency. Enjoy, and enjoy LAC in Graz this year Frank ________________________________ QMidiArp is an advanced MIDI arpeggiator, programmable step sequencer and LFO. Everything is on http://qmidiarp.sourceforge.net qmidiarp-0.5.2 (2013-05-09) New Features o Tempo changes are again possible while running, both manually or by a Jack Transport Master o Jack Transport position is now taken into account when starting, QMidiArp used to start always at zero o Muting and sequencer parameter changes can be deferred to pattern end using a new toolbutton o Modules in the Global Storage window have mute/defer buttons o Global Storage location switches can be set to affect only the pattern o Non Session Manager support with "switch" capability (thanks to Roy Vegard Ovesen) General Changes o NSM support requires liblo development headers (liblo-dev package)

10 years, 6 months

OpenAV Productions : Sorcer

by Harry van Haaren

Hey Everybody, I'm happy to announce OpenAV productions: http://openavproductions.com OpenAV productions is a label under which I intend to release my linux-audio software projects. The focus of the software is on the workflow of creating live-electronic music and video. The release system for OpenAV productions is one based on donations and time, details are available on http://openavproductions.com/support Sorcer is a wavetable synth, and is ready for release. Check out the interface and demo reel on http://openavproductions.com/sorcer Greetings from the LAC, -Harry

10 years, 11 months

itop

by elboulangero

Hello everyone, lately I had to fight big XRUN troubles, and thanks to this forum I finally solved that. This excellent thread saved me: http://linuxaudio.org/mailarchive/lau/2012/9/5/192706 On my long quest, I tried to see a little bit more what happened with the IRQs on my system. I searched for a kind of 'top' utility to monitor the interrupts, but the only apps I found were either deprecated, or missed some cool features. So, I ended up writing my own tool to monitor the file /proc/interrupts. It's available a this address: https://gitorious.org/elboulangero/itop As its name indicates, it behaves pretty much like top, but for interrupts. It's quite a simple thing, that I tried to enhance a bit with some cool features: + refresh period can be specified. + two display modes: display interrupts for every CPU, or only a sum of all CPU. + display every interrupt (sorted like /proc/interrupts), or only active interrupts (sorted by activity). + in case the number of interrupts changes during the execution of itop (due to a rmmod/modprobe), it's handled without any fuss. + command-line options are also available as hotkeys for convenience. + at last, the program display a summary on exit. The idea is that this summary could be copied/pasted in emails to help debugging. If anyone is interested, feel free to try and comment ! Cheers

10 years, 11 months

Support for Steinberg UR22 (Yamaha USB chipset 0499:1509)?

by cullinan＠rocketmail.com

Hi, there! I just got a Steinberg UR22 USB<->Audio / MIDI interface for my semi-prof. DJ work. I am working on Linux (only), but wanted to give this new product a chance. Well, it's basically a nice device but with an Yamaha chipset in it which seem unfortunately not supported from the latest (3.7.9) linux kernel(s). My questions are: - How likely is it to get this device fully supported on linux? - What needs to be done to get this device fully supported? I am an electrical engineer and did some linux device driver development in the past on some DMA stuff, but I am quite new to linux audio. But I am willing to help somewhere... I could try to reverse-engineer the USB stuff on Windows with the usbmon/usbsnoop/libusb things... But if it's too much work, to get it working, I could also just decide to send it back to Steinberg and thank them for not supporting Linux with their products. The USB information I could get is: $ aplay -l doesn't list that device $ lsusb ... Bus 002 Device 016: ID 0499:1509 Yamaha Corp. ... (See the more comprehensive $ lsusb -v below...) Related USB devices are: 0499:1501 Yamaha 0499:1507 Yamaha THR10 0499:1509 Yamaha Steinberg UR22 (what I am talking about) 0499:150c Yamaha THR10C Related search resuls are: Getting a Steinberg CI1 USB audio interface to work http://ubuntuforums.org/showthread.php?t=1696336 Yamaha THR10 amp / USB2.0 audio interface http://linux-audio.4202.n7.nabble.com/Yamaha-THR10-amp-USB2-0-audio-interfa… Thank you, Clemens (different one as in last link) -----8<----- $ lsusb -v -d 0499:1509 Bus 002 Device 014: ID 0499:1509 Yamaha Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0499 Yamaha Corp. idProduct 0x1509 bcdDevice 1.00 iManufacturer 1 Yamaha Corporation iProduct 2 Steinberg UR22 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 442 bNumInterfaces 5 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 0a 24 01 00 01 34 00 02 01 02 ** UNRECOGNIZED: 0c 24 02 01 01 01 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 02 01 03 00 01 00 ** UNRECOGNIZED: 0c 24 02 03 03 06 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 04 01 01 00 03 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 01 01 00 ** UNRECOGNIZED: 0e 24 02 01 02 03 18 02 44 ac 00 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x002a 1x 42 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 01 01 00 ** UNRECOGNIZED: 0e 24 02 01 02 03 18 02 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x004e 1x 78 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 01 01 00 ** UNRECOGNIZED: 0e 24 02 01 02 03 18 02 10 b1 02 00 ee 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0096 1x 150 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 04 01 01 00 ** UNRECOGNIZED: 0e 24 02 01 02 03 18 02 44 ac 00 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x002a 1x 42 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 04 01 01 00 ** UNRECOGNIZED: 0e 24 02 01 02 03 18 02 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x004e 1x 78 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 04 01 01 00 ** UNRECOGNIZED: 0e 24 02 01 02 03 18 02 10 b1 02 00 ee 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0096 1x 150 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 24 00 ** UNRECOGNIZED: 06 24 02 02 01 00 ** UNRECOGNIZED: 09 24 03 02 01 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 255 Vendor Specific Protocol iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 24 00 ** UNRECOGNIZED: 06 24 02 02 01 00 ** UNRECOGNIZED: 09 24 03 02 01 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0000 (Bus Powered) -----8<-----

10 years, 11 months

Alsa works fine the default output but distorts/stutters with hw:0/plughw:0

by Muffinman

Hello all, I'm trying to add Alsa support to an application with mixed results. When using the default audio output it sounds fine (my dac display 48KHz as sampling rate, which doesn't equal the source signal of 44.1) but when using hw:0 it is heavily distorted and with plughw:0 the sound stutters at a steady interval of about 2Hz (in these last two cases my dac display 44.1kHz as the sampling rate). When using libao plughw:0 sound fine (and my dac always display 44.1kHz). Can anyone give me a direction (or answer) where I might go wrong here. Unfortunately I'm a bit out of my league here and I do not yet understand the data that fills buf[]. However, I do know that the data it supplies works fine with libao, so I do suspect it is something with the code below. Thanks in advance, Maarten ps. it is supposed to add Alsa to the following: https://github.com/abrasive/shairport/tree/1.0-dev static void start(int sample_rate) { if (sample_rate != 44100) die("Unexpected sample rate!"); int ret, dir = 0; snd_pcm_uframes_t frames = 32; ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0); if (ret < 0) die("Alsa initialization failed: unable to open pcm device: %s\n", snd_strerror(ret)); snd_pcm_hw_params_alloca(&alsa_params); snd_pcm_hw_params_any(alsa_handle, alsa_params); snd_pcm_hw_params_set_access(alsa_handle, alsa_params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(alsa_handle, alsa_params, SND_PCM_FORMAT_S16); snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2); snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, (unsigned int *)&sample_rate, &dir); snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &frames, &dir); ret = snd_pcm_hw_params(alsa_handle, alsa_params); if (ret < 0) die("unable to set hw parameters: %s\n", snd_strerror(ret)); } static void play(short buf[], int samples) { int err = snd_pcm_writei(alsa_handle, (char*)buf, samples); if (err < 0) err = snd_pcm_recover(alsa_handle, err, 0); if (err < 0) die("Failed to write to PCM device: %s\n", snd_strerror(err)); }

11 years, 8 months

Re: [LAD] Open-source Band-in-a-Box replacement

by Michael Niemeck

Quoting conrad berhörster <beat.siegel.vier(a)gmx.de>: > maybe this helps > http://welltemperedstudio.wordpress.com/code/lemma/ lemma looks cool. I will definitely give it a try as soon as I get my daw back from the repair shop... > and take a look at > impro-visor > http://www.cs.hmc.edu/~keller/jazz/improvisor/ I also worked with impro-visor a lot. Basically it has too many features for my requirements. I just want an easy-to-use tool to provide a playback track for practice sessions for those of us that lack the skill or the extra hands to accompany themselves on the piano during practice. I remember the playback quality to be very good, though. I never got to use all the melody-centered functionalities, it's a different use-case really. That's also one of my design goals - focus on the central requirement, keep everything out that's not absolutely necessary, make it as easy to use as possible, and in the end really focus on the quality of the output (i.e. the music that can be heard). > And maybe you can explain a little bit about your ideas about the > pattern-less approach Happy to, though it might get a bit longish from here on. Bear with me. My reasoning goes like this: Take the bass part of a very simplistic straight 4-beat groove. In a fictional pseudo-pattern-defining-format you may specify something like this: -) Play the bass note of the chord on beat 1. -) Play the bass note of the chord on the off-beat after 2. -) Play the bass note of the chord on beat 3. -) Play the bass note of the chord on the off-beat after 4. This will sound fine on straight-forward one-chord-per-bar tunes. Even if you change chord every two beats it will sound reasonably good. But what of bars with a chord change on every beat, such as happens often in Jazz, especially in turnarounds and the like? The bass player will miss every other chord! Sure, you could add a rule like -) In addition, play the bass note of the chord at every chord change. assuming your pseudo-pattern-defining-format allows this. But then the notes on the off-beats defined earlier will not sound too good. A real life bass player would in such turnaround bars probably play a single note on every chord change and leave out the off-beat notes. Or consider off-beat chord changes, also a common thing. Just think about what a typical pattern definition will make of them, and what a real player would play. Still, it's possible to express all this in a pattern definition. But always provided the engine reading the pattern files supports it! And I think that sooner rather than later you reach the point where understanding the syntax of a sufficiently powerful pattern format and actually writing good patterns in it is not that much easier than programming in any well-structured contemporary programming language. Bottom line: I think everyone able to define patterns so complex that they actually sound really good is just as able to program them given a well-structured framework to work in. What you gain by actually programming a groove is that you are no longer limited by the abilities of the pattern interpreter, but have the full expressive power of the programming language at your disposal. Random elements, variations in timing and volume to make it sound more human, even the occasional wrong note, all that is possible without any changes to the engine itself. BTW, the same holds for the voicing of chords. Consider the ugly but simple progression D7 C9 Bb6. By defining for the bass to -) Play the root note of the chord in octave #3 will give you a not only ugly but also quite unrealistic jump of a seventh up at the last change. It gets worse if the voicing of the pianists right hand just builds the chords up from the root. Playing the above progression with all the chords in root position will not only violate a lot of voicing rules, it will also sound pretty crappy. A real-life pianist would probably play all three chords with the D at the bottom, or maybe even move up through D E F or something similar. Actually programming such rules is doable, defining them in a declarative fashion is much much harder. So how does this actually work? At the lowest (and most convenient) level you just define your Groove as a number of Players, each representing typically one instrument. In your Player you have two hook methods, one that gets called once for every bar and one that gets called once for every chord change. Within these methods you define (actually you program) what notes should be added to the final music. For that you have not only methods to actually add notes to the track, but also a (hopefully still growing) number of convenience methods, for example to get the proper notes for the current chord from a voicing engine or to get not only the current chord but also the previous and next chord(s) (for example to program cool walking bass lines you need the full context) as well as their suggested voicings, etc. In addition you can do basically everything that can be programmed. In the most simple case, programming in that layer is not harder than defining patterns. The simple pattern from above would look something like this: void createEvents(Bar bar) { addNote(getVoicing(getChordAt(0.0)).get(0), 0.0, 1.0, 0.35); addNote(getVoicing(getChordAt(0.375)).get(0), 0.375, 1.0, 0.12); addNote(getVoicing(getChordAt(0.5)).get(0), 0.5, 1.0, 0.35); addNote(getVoicing(getChordAt(0.875)).get(0), 0.875, 1.0, 0.12); } The difference is that you are not limited anymore. You can even skip the convenience layer and directly implement your own Player object, in which you receive a list of music information (Bars, Chord changes, Volume changes, Tempo changes), an assigned MIDI channel number and the set MIDI resolution, and must return simply an arbitrary list of MIDI events. To everyone who made it this far: Thanks for listening ;-) Mike -- Michael Niemeck Krausegasse 4-6/3/6 1110 Wien michael(a)niemeck.org +43 1 9417017 +43 660 9417017 ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

11 years, 9 months

Open-source Band-in-a-Box replacement

by Michael Niemeck

No, not the umpteenth person looking for one. I want to actually set out to try and build one. I'm still in the early stages of scoping, architecturing and prototyping (read: not ready to share anything yet), but still there's a few questions I would like to get feedback on: 1) Would there be any demand for such an open source initiative at all, or is everyone seriously needing this type of application using The Real Thing anyway? 2) Would anyone be interested in collaborating? The core and GUI code are not really suitable for collaborative development (yet), and anyway not the real issue (although I'm open to any suggestion and/or wishes regarding the user experience). What the project _does_ need however in order to get serious at some point, is a lot of brains put into the actual music creation routines, which I'm designing from the start to be easily contributed to. It's in Java and I'm trying to put in as much convenience methods as possible, so a lot of programming know-how is not really necessary to contribute. What I'd rather need is people with a lot of musical know-how, like voicing theory or in-depth style knowledge, and just enough formal thinking to be able to put their ideas into some form of algorithms (if Java is a problem I'm willing to accept any for of pseudo-code as well...). 3) Is this the right place to ask? (Probably should have put this as 1)...). Seriously, if you have any suggestions where else I might put this up, please let me know as well. If there are some 'yes's out there, I'll be happy to post more information. Cheers Mike ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

11 years, 9 months

Re: [LAD] 4-6 or 8 cores ??

by Jonathan Woithe

Hi Geoff On Sun, 16 Jun 2013 20:49:26 +1000 you wrote: > Going to finally build a new machine. I'ts going to be Intel this time - > AMD for 15 years or so - can any one here give some advice as to how > many cores are optimal given current kernel >3.8 performance. Any > install/operational issues? Any pitfalls ? I can't provide much in the way of scientific evidence; there are others who know far more about the technical realities of this - and push their machines much harder - than I do. Personally, I have always run with the idea that so long as one has N independent processes whose links to other processes are limited to the consumption or production of streams, then one could theoretically max out N CPU cores. However in most practical cases involving audio and video one is running the streams at real world speed, and this obviously limits the extent that each process requires a CPU. The only time a single process would have the chance to max out a single CPU core would be when freewheeling with jack for example, or doing a final render of a video. If one spends most of the time interacting with their AV software, this means that there's no simple answer to the "how many cores are optimal" questions. It depends on the precise mix of software you're running, what each process requires of a CPU core, how much I/O each process instigates, and so on. Another caveat is jackd: my current understanding is that jack2 can better utilise multiple cores, but I'm happy to be corrected on this point by anyone with more knowledge than I (I really haven't looked into this recently because for my current situation it's academic). My system has been based on a first-generation i7 for the last couple of years and I've noticed no major issues. However, in terms of audio work I'm not really pushing the system all that hard during real work (I don't have soft-synths running generally, and the plugins I use tend to be fairly frugal with CPU requirements). This gives me 4 cores with hypertheading and when I've done tests to see what it could handle, an audio-like workload is able to push well above the 400% loading (so the hyperthreading seems to be doing something useful). Having said all that and knowing the sort of work you do, I would probably err on the side of getting as many cores as you can reasonably afford. As time goes on they won't go astray; you'll have the flexibility to experiment with new ways of doing things without being too constrained by the number of cores at your disposal. A final comment is that with the release of Intel's Haswell-based CPUs we are at an interesting point in time. These new cores are certainly a big win for mobile computing due to their lower power consumption for a given performance level. However, whether the increase in outright performance - the primary metric for a desktop - justifies the "new product" premium these will attract for the next 6-12 months remains to be seen, especially since one would also expect some runout discounting on the 4th generation CPUs in the coming months. Regards jonathan

11 years, 9 months

A3 clicks

by Fons Adriaensen

Anyone else noticed this: Ardour 3.1-3-g1606996: changing the meter source (in/pre/post/custom) in a mixer strip generates clicks in the audio output. ?? Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

11 years, 9 months

error with jalv

by Gianfranco Ceccolini

hi LADs I installed linux on a new macbook retina and have getting some trouble running jalv. it gives me the error gian@gian-MacBookPro:~/mod/LV2/jalv-1.4.0$ jalv.gtk http://guitarix.sourceforge.net/plugins/gx_amp#GUITARIX Plugin: http://guitarix.sourceforge.net/plugins/gx_amp#GUITARIX UI: http://guitarix.sourceforge.net/plugins/gx_amp#gui JACK Name: GxAmplifier-X Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jackdmp 1.9.10 Copyright 2001-2005 Paul Davis and others. Copyright 2004-2013 Grame. jackdmp comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details no message buffer overruns no message buffer overruns no message buffer overruns JACK server starting in realtime mode with priority 10 audio_reservation_init Acquire audio card Audio1 creating alsa driver ... hw:1,0|hw:1,0|256|2|44100|0|0|nomon|swmeter|-|32bit configuring for 44100Hz, period = 256 frames (5.8 ms), buffer = 2 periods ALSA: final selected sample format for capture: 24bit little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 24bit little-endian ALSA: use 2 periods for playback Block length: 256 frames MIDI buffers: 32768 bytes Comm buffers: 131072 bytes Update rate: 2 Hz using block size: 256 MasterGain = -15,000000 PreGain = 0,000000 Distortion = 20,000000 Drive = 0,250000 Middle = 0,500000 Bass = 0,500000 Treble = 0,500000 Cabinet = 10,000000 Presence = 5,000000 model = 0,000000 t_model = 1,000000 c_model = 0,000000 Inconsistency detected by ld.so: dl-open.c: 684: _dl_open: Assertion `_dl_debug_initialize (0, args.nsid)->r_state == RT_CONSISTENT' failed! JackEngine::XRun: client = GxAmplifier-X was not finished, state = Triggered JackAudioDriver::ProcessGraphAsyncMaster: Process error gian@gian-MacBookPro:~/mod/LV2/jalv-1.4.0$ Unknown error... terminate called after throwing an instance of 'Jack::JackTemporaryException' what(): due to the new retina display I've installed both KDE and GNOME. I wonder if the problems com from that. Any help is appreciated. kind regards Gian

11 years, 9 months

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Linux-audio-dev June 2013